Various forms of communications can be performed in packet-based networks, such as electronic mail, web browsing, file transfer, and so forth. With the increased capacity and reliability of packet-based networks, voice communications (along with other forms of real-time, interactive communications) have also become feasible. In such communications, voice and other real-time data are carried in packets that are sent across the network.
Various standards have been proposed for voice and multimedia communications over packet-based networks. One such standard is the H.323 Recommendation from the International Telecommunication Union (ITU). Another standard for voice and multimedia communications is the Session Initiation Protocol (SIP), as developed by the Internet Engineering Task Force (IETF). Generally, H.323, SIP, and other control protocols are used for negotiating session information to coordinate the establishment of a call session. Once negotiation setup has been completed, packetized media (including voice or other forms of real-time data) can flow between endpoints. A media transport protocol, such as the Real-Time Protocol (RTP), is used for conveying packetized media between the endpoints.
Because communications, such as call sessions, often traverse a public network such as the Internet, security is a concern. However, security must be balanced against the needs of users for convenient access of various types of services, such as electronic mail, web browsing, and telephony communications. A need thus continues to exist for improved methods and apparatus for enabling robust secure communications over various networks.